As is well known, it is possible to save bandwidth, and thus reduce transmission costs, in digital telecommunications systems by using so-called speech coding. Such coding exploits redundancies in the speech signals to compress the signals from their original rate of (typically) 64 kilobits/second (kbps) to some lower rate prior to transmission. In current practice, for example, the most-widely used speech coding algorithm is adaptive differential pulse code modulation (ADPCM) which is used to compress 64 kbps speech to 32 kbps speech.
In the meantime, the past decade or so has seen considerable progress in the speech coding art to the point where it has been shown that the 64 kbps speech can be compressed to a rate as low as 2.4 kbps. Although these newer techniques have not found widespread commercial use as of yet, commercial interest therein has increased significantly in the last few years. One of the factors contributing to this increased level of commercial interest is the fact that today's speech compression algorithms are able to achieve greater levels of compression at more acceptable levels of distortion and signal delay than have been achieved in the past. Another such factor is the accelerating pace of the conversion of the installed base of telecommunications equipment and facilities, both in this country and around the world, from analog to digital technologies.
Inherent in virtually any speech coding algorithm is the loss of a certain amount of speech information. Thus the re-constituted signal arrived at upon performing the inverse, decoding, process is somewhat distorted relative to the original. It is also somewhat delayed relative thereto. In general, the level of this distortion and delay increases as the level of compression increases. Moreover, if a speech signal is subjected to two or more encoding/decoding cycles, each cycle adds its own measure of distortion and delay to the signal being processed.
This has not proved to be a problem with, for example, 64-to -32 kbps ADPCM coding of speech signals because the levels of distortion and delay introduced therein are quite small. On the other hand, most of the newer speech coding algorithms, when operated at, say, a 64-to-16 kbps level of compression, can withstand perhaps only one encoding/decoding cycle before noticeable, and therefore objectionable, distortion and/or delay occur. This is a significant limitation because, in many telecommunications environments, the connection established between two communicating endpoints may include two or more pairs, or "tandems," of encoder/decoders, or "codecs"--the units which actually carry out the speech encoding and decoding. This can happen, for example, when a connection set up within a private telecommunications network includes tow or more PBXs. It is also inherent in a mobile-station-to-mobile-station connection in a digital cellular radio system.
It thus appears that the extent to which the advantages afforded by the high-compression speech coding algorithms that have been developed in recent years can be exploited will hinge to a significant extent on the ability to minimize, or avoid, the distortion and delay which may be introduced when the speech signal is routed through two or more codec tandems.